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Unit - 4
Modulation is the addition of information (or the signal) to an
electronic or optical signal carrier. Modulation can be applied to direct
current (mainly by turning it on and off), to alternating current, and to
optical signals. For most of radio and telecommunication today, the
carrier is alternating current (AC) in a given range of frequencies.
Common modulation methods include:
Amplitude modulation (AM), in which the voltage applied to the carrier
is varied over time
Frequency modulation (FM), in which the frequency of the carrier
waveform
is
varied
in
small
but
meaningful amounts
Phase modulation (PM), in which the natural flow of the alternating
current waveform is delayed temporarily
These are sometimes known as continuous wave modulation methods to distinguish them from pulse
code modulation (PCM), which is used to encode both digital and analog information in a binary way.
Radio and television broadcast stations typically use AM or FM. Most two-way radios use FM, although
some employ a mode known as single sideband (SSB).
The transmission of analog data or speech which is in continuous form is known
as pulse modulation.
At some certain levels or points, the wave formation can be seen in a pulse
modulation system. In this synchronizing, pulses are sent with the information
related to the signal at different time samples. There are two main categories in
which pulse modulation can be divided.
1. Analog pulse modulation
2. Digital pulse modulation
Modem Modulation and Demodulation
A computer with an online or Internet connection that connects over a regular analog phone line
includes a modem. This term is derived by combining beginning letters from the words modulator and
demodulator. In a modem, the modulation process involves the conversion of the digital computer
signals (high and low, or logic 1 and 0 states) to analog audio-frequency (AF) tones. Digital highs are
converted to a tone having a certain constant pitch; digital lows are converted to a tone having a
different constant pitch. These states alternate so rapidly that, if you listen to the output of a computer
modem, it sounds like a hiss or roar. The demodulation process converts the audio tones back into
digital signals that a computer can understand directly.
Modulation is the process of varying one or more properties of a periodic waveform, called the carrier
signal (High Frequency Signal), with a modulating signal which typically contains information to be
transmitted.
In telecommunications, modulation is the process of conveying a message signal, for example a digital
bit stream or an analog audio signal, inside another signal that can be physically transmitted.
Modulation of a sine waveform is used to transform a baseband message signal into a passband signal.
A device that performs modulation is known as a modulator and a device that performs the inverse
operation of modulation is known as a demodulator (sometimes detector or demod). A device that can
do both operations is a modem (from "modulator–demodulator").
PCM is a method of converting an analog into digital signals using quantization by
amplitude and quantization by time Information which mean digitalization of the
analog signal. Since signal in an analog form cannot be processed by digital
computers so it's necessary to convert them into digital form. Digital data can be
transported robustly over long distances unlike the analog data and can be
interleaved with other digital data so various combinations of transmission
channels can be used.
The range of values which the signal can achieve (quantization range) is divided
into segments; each segment has a segment representative of the quantization
level which lies in the middle of the segment. To every quantization segment (and
quantization level) one and unique code word (stream of bits) is assigned. The
value that a signal has in certain time is called a sample. The process of taking
samples is called quantization by time. After quantization by time, it is necessary
to conduct quantization by amplitude. Quantization by amplitude means that
according to the amplitude of sample one quantization segment is chosen (every
quantization segment contains an interval of amplitudes) and then record
segments code word.
Companding refers to a technique for compressing and then expanding (or
decompressing) an analog or digital signal. It is a combination of the words
"compressing" and "expanding."
This twin-sequential process is non-linear overall but linear over short periods of
time. Data is compressed before being transmitted. Then, it is expanded at the
receiving end using the same non-linear scale to restore it to its original form, but
with reduced noise and crosstalk levels (meaning reduced disruption of, or
interference with, signals in an adjacent circuit). This disruption or interference is
commonly from alternating current (AC), direct current (DC) or other transmission
lines.
Companding is used as a complement to the process of modulation and
demodulation. In this process a voice signal is compressed, then changed from
analog to digital, transmitted and converted back from digital to analog before it
is expanded again.
For audio analog signals, the amplitude of weak signals is raised and the amplitude of strong signals is
decreased, thereby altering (compressing and expanding) the dynamic range of the signals. The
technique is used in AM, FM and single-sideband modulation radio and is helpful in improving the
quality of amplified voice and musical instrument sounds. Dolby and dbx noise reduction also employ
companding. Concert audio systems and noise reduction technologies such as dbx and Dolby use a
triplet of amplifiers to accomplish this process, meaning a logarithmic amplifier, a variable-gain linear
amplifier and an exponential amplifier.
For digital audio signals, companding is used in pulse code modulation (PCM). The process involves
decreasing the number of bits used to record the strongest (loudest) signals. In the digital file format,
companding improves the signal-to-noise ratio at reduced bit rates. For example, a 16-bit PCM signal
may be converted to an eight-bit ".wav" or ".au" file.
Another application of companding involves professional wireless microphones, which have a larger
dynamic range than is possible through radio transmission. By decreasing the amplitude of signals, the
signals may be transmitted and then expanded at the receiver, where the original signals are
reproduced by the receiving electronic equipment.
Mu-Law is the standard codec (compression/decompression) algorithm for pulse
code modulation (PCM) from the CCITT (Consultative Committee for International
Telephone and Telegraph). A companding (compression/expanding) method, muLaw makes it possible to improve the signal-to-noise ratio without requiring the
addition of more data. Mu-Law, as a term, was derived from µ-Law, because the
Greek letter µ is pronounced myoo. The term is sometimes seen as u-Law,
although the pronunciation remains the same.
Mu-Law is used in the United States and Japan. The other type of codec
algorithm, A-Law, is the standard used in Europe and elsewhere.
• The -law algorithm (μ-law) is a companding algorithm, primarily used in
the digital telecommunication systems of North America and Japan. Its
purpose is to reduce the dynamic range of an audio signal. In the analog
domain, this can increase the signal to noise ratio achieved during
transmission, and in the digital domain, it can reduce the quantization error
(hence increasing signal to quantization noise ratio).
A-Law is the standard codec (compression/decompression) algorithm for pulse
code modulation (PCM) from the ITU-T (the Telecommunication Standardization
Sector of the International Telecommunications Union). A-Law is the type of PCM
used in most of the world. The other type, mu-Law, is used in the United States
and Japan.
•
•
A-law algorithm used in the rest of worlds.
A-law algorithm provides a slightly larger dynamic range than the mu-law at the cost of worse
proportional distortion for small signals. By convention, A-law is used for an international
connection if at least one country uses it.
In simpler terms, this means that sound is represented as a wave, and humans can only hear audio in
the middle of the wave. We can remove data from the upper and lower frequencies of a sound, and
humans will not be able to hear a significant difference. Both Mu-Law and A-Law take advantage of this,
and are able to compress 16-bit audio in a manner acceptable to human ears. A-Law and Mu-Law
compression appear to have been developed at around the same time, and basically only differ by the
particular logarithmic function used to determine the translation. When we get to the work of
implementing the algorithms, you will see that the differences are nominal. The main difference is that
Mu-Law attempts to keep the top five bits of precision, and uses a logarithmic function to determine
the bottom three bits, while A-Law compression keeps the top four bits and uses the logarithmic
function to figure out the bottom four. Both of these algorithms are used as telecommunication
standards, A-Law being used mainly in Europe, and Mu-Law being used in the United States.
T1
T1 is a special type of fiber optic telephone line and it was developed by AT&T Bell Labs. T1 is the most
commonly used digital transmission service in the United States. T1 line is capable of transferring the
broadband digital data at very high speed i.e. 1.54 Mbps. T1 is an expensive solution for data
transmission as compared to the regular telephone lines. Currently T1 is not cost effective for the home
users.
A large number of the businesses in USA, Canada and Japan use T1 lines to connect to the internet.
There are different ways to provide T1 transmission such as twisted pair cables, coaxial cable modems,
fiber optic systems, common careers and digital radios.
In large networks T1 signals can be boosted up to 100 miles by using the repeaters. In
telecommunication terminology, T1 is also known as DS1 and T1/DS1 is a method of connecting the
digital communication systems with the telecommunication industry and North America.
E1
E1 is a digital data communication system for the European data transmission format. E1 carries the
data signals at the speed of 2 Mbps (Full duplex). E1 and T1 lines can be interconnected with each other
for international data transmission.
E1/T1 are used for the leased lines transmission. E1 is ideal for the voice traffic and it can carry 32 voice
conversions. An E1 line 32 64-Kbps channels and each channel may be used to send and receive data
and voice.
E1/PR1 (Primary rate interface) supports 30 B channels and one D channel. Primary Rate Interface (PRI)
configurations are used to receive multiple analog calls from the dial-in traffic and analog-modems.
Line Coding Basics
Transmission of serial data over any distance, be it a twisted pair, fiber optic link,
coaxial cable, etc., requires maintenance of the data as it is transmitted through
repeaters, echo chancellors and other electronically equipment. The data
integrity must be maintained through data reconstruction, with proper timing,
and retransmitted. Line codes were created to facilitate this maintenance. In
selecting a particular line coding scheme some considerations must be made, as
not all line codes adequately provide the all important synchronization between
transmitter and receiver. Other considerations for line code selection are noise
and interference levels, error detection and error checking, implementation
requirements, and the available bandwidth.
AMI (Alternate Mark Inversion)
It is a polarity inversion in which polarity of each pulse is inverted with regards to
the preceding pulse.
The data bits are transmitted to the line as pulses representing "1" and spaces (no
pulse) representing "0". The nominal "1" pulse (mark) voltage is 3V for T1 and 120
ohm E1 and 2.37 V for 75 ohm E1.
If pulses had only one polarity - that would introduce a DC component to the line.
The signal having a DC component can't be transmitted, because the repeaters
placed along the line to retransmit the signal, have DC power feeding supplied on
the same line. To overcome this problem the polarity of each pulse is inverted
with regards to the preceding pulse. This polarity inversion is called AMI
(Alternate Mark Inversion) signal (or line) coding. Such bipolar pattern also halves
the fundamental frequency of the signal which results in less attenuation and
group delay. The AMI signal coding does not address another problem, though.
When there is long sequence of zeros the equipment at the other end of the line
can't synchronize and thinks that the link is lost. To solve this problem pulses are
inserted in each sequence of 8 (in T1) or 4 (E1) continuous zeros. Such "artificial"
pulse has the polarity opposite to what is required by the AMI rule.
Used in North American T1 (1.544MHz) and T1C (3.152MHz) lines.
HDB3 (High Density Bipolar 3)
Another coding scheme is HDB3, high density bipolar 3, used primarily in Europe
for 2.048MHz (E1) carriers. This code is similar to BNZS in that it substitutes
bipolar code for 4 consecutive zeros according to the following rules:
If the polarity of the immediate preceding pulse is (-) and there have been an odd
(even) number of logic 1 pulses since the last substitution, each group of 4
consecutive zeros is coded as 000-(+00+).
If the polarity of the immediate preceding pulse is (+) then the substitution is
000+(-00-) for odd (even) number of logic 1 pulses since the last substitution.
B8ZS (Bipolar with 8 Zero Substitution)
B8ZS is used commonly in North American T1 (1.544MHz) and T1C (3.152MHz)
carriers. For every string of 8 zeros, bipolar code is substituted according to the
following rules:
If the immediate preceding pulse is of (-) polarity, then code each group of 8 zeros
as 000-+0+-.
If the immediate preceding pulse is of (z) polarity, then code each group of 8
zeros as 000+-0-+.
SDH is very useful equipment that is used for the telecommunication sector in easy transfer of data.
Earlier PDH was widely used but due to some of its weaknesses, SDH has replaced the use of PDH. But
not everywhere. Point to Point applications are still used mainly by PDH, and also, it's cheaper.
PDH or the plesiochronous digital hierarchy is a popular technology that is widely
used in the networks of telecommunication in order to transport the huge
amounts of data over the digital equipment for transportation like microwave
radio or fiber optic systems. The term plesiochronous has been derived from the
Greek word ‘plesio’ that means ‘near’ and ‘chronos’ meaning time. This means
that the PDH works in a state when the various different parts of a network are
clearly synchronized. But with the change in technology, the PDH is now being
replaced by the SDH or what is popularly called as synchronous digital hierarchy.
The SDH is useful equipment that is used in most of the telecommunications
networks.
This PDH helps in proper transmission of the data that generally runs at the
similar rate but allows some slight variation in the speed than the nominal rate.
The basic transfer rate of the data is 2048 kilobits per second. For instance, in
each speech transmission, the normal rate breaks into different thirty channels of
64 kilobits per second along with two different 64 kilobits per second in order to
perform the tasks of synchronization and signaling. The typical rate of
transmitting the data over the fiber optic systems is 565 Mbit/sec in order to
transport the data in the long distance. But as the technology has improved with
the passing of time, now the telecommunication companies have replaced the
PDH equipment with that of the SDH equipment, which has the capability of
transmitting the data at much higher rates as compared to the PDH system.
The SDH is an international standard that is highly popular and used for its high
speed data transfer of the telecommunication and digital signals. This
synchronous system has been specially designed in order to provide a simple and
flexible network infrastructure. This system has brought a considerable amount of
change in the telecommunication networks that were based on the optical fibers
as far as performance and cost were concerned.
The weaknesses that PDH faced paved way for the introduction and use of the
SDH systems. Although the PDH proved to be a breakthrough in the field of digital
transmission, the weaknesses that made it less demanded includes:
 Asynchronous structure that is rigid.
 Restricted management capacity.
 Non availability of world standard on the digital formats.
 No optical interfaces world standard and without an optical level,
networking is not possible.
But if we compare the PDH system with that of the SDH system, the latter one has a large number of
advantages. Some of the most common advantages enjoyed by the usage of SDH include:
 optical interfaces
 capability of powerful management
 world standard digital format
 synchronous structure is flexible
 cost effective and easy traffic cross connection capacity and add and drop facility
 reduced networking cost due to the transversal compatibility
 forward and backward compatibility
Apart from all the advantages mentioned above, the SDH also has various management capabilities such
as performance management, security and access management, configuration management and the
event or the alarm management. So, we can clearly make a distinction between the PDH and SDH
systems so that as per the needs of the telecommunication, the appropriate transmission system can be
used.
Synchronous Optical Networking (SONET)
Synchronous optical networking is a standardized digital communication protocol
that is used to transmit a large volume of data over relatively long distances using
a fiber optic medium. With SONET, multiple digital data streams are transferred at
the same time over optical fiber using LEDs and Laser beams. SONET is a product
of the American National Standards Institute (ANSI).